Method, apparatus, and system for analysis, evaluation, measurement and control of audio dynamics processing

ABSTRACT

A method, apparatus, and system for measuring and analyzing the effects of dynamics modifying processors on a signal. This new approach utilizes statistical analysis techniques to provide a direct comparison and evaluation between the processed signal and the unprocessed signal&#39;s dynamic characteristics. The method identifies and quantifies Effective Dynamic Range, Clip Tolerance, Lower Limit Tolerance, Crest Factor, and Diminuendo Factor, using either peak or r.m.s values. In an alternative embodiment, the invention allows for user adjustment and control of the relative relationship of Crest Factor and Diminuendo Factor, which the user may perceive as loudness.

CROSS REFERENCE TO RELATED APPLICATIONS

This is a Continuation of U.S. Utility patent application Ser. No.14/095,951, filed Dec. 3, 2013.

FIELD OF INVENTION

Generally, this invention relates to the class of inventions known asElectrical Audio Signal Processing Systems and Devices. Morespecifically, this invention belongs to the sub-classes of inventionsrelating to Monitoring and Measuring of Audio Devices and AmplitudeCompression and Expansion.

BACKGROUND OF INVENTION

Audio reproduction has encountered and wrestled with the twin problemsof maximizing dynamic range and minimizing compression since thepatenting of the electro-dynamic loudspeaker by Peter Jensen in 1927.When addressing the goal of maximum fidelity to the original soundsource, audio reproduction systems elements such as amplifiers andloudspeakers must aim to provide a dynamic range which equals or exceedsthat of the original source signals. Where the selected availabletechnologies do not allow sufficient dynamic range and linearity to beachieved in the reproduction, system dynamic range is lost either orboth at the system minimum capability (the electronics noise floor orminimum motive force in electro-acoustic elements), or at the systemmaximum capability (clipping or overload). In loudspeakers, systemoverload can be evident in a lessening of the ratio of output to inputas level nears maximum as the system looses linearity such that anincrease in input is only partially reflected in an increase in output.This effect in loudspeakers is commonly referred to as powercompression. A similar effect, known as signal compression, being areduction in through gain, the ratio of output to input levels, can bedeliberately introduced as a signal processing element in the electronicsignal path. This latter technique has been widely used to amelioratethe deleterious effects of reproduction system overload and to allow thesource dynamic range to be reduced to allow the signal program materialto be reproduced without gross overload of the available sound system.As an example, a typical FM radio transmission may achieve a dynamicrange on the order of 50 dB. In contrast, in a typical concert hall witha background noise level of 35 dBA performance levels can reach over 115dBA, a dynamic range of 80 dB. The performance dynamic range exceeds thedynamic range available for an FM broadcast of the concert by 30 dB.Signal dynamic range compression must be imposed on the source signal toreduce the dynamic range to that available in the reproduction medium.An adjunct dynamic range modifier is an expander which takes thecompressed signal and attempts to increase the dynamic range to moreclosely mimic the original source signal. An audio system that hassubstantial compression will typically have poor dynamic range. Athigher levels, compression becomes objectionable to most listeners, andis a cue that the system performs poorly.

The typical audio amplifier is a voltage-controlled device. A typicalaudio amplifier will take a small time-varying input voltage (typically50 mV peak) and will output an identical, though larger, version of thetime-varying signal. The overall ratio of the output-to-input is theeffective gain of the system, which is often expressed in decibels (dB).When a user increases the volume, they are, in reality, increasing thevoltage output from a voltage-controlled amplifier. Thevoltage-controlled amplifier is limited by its voltage rail, which is,essentially, the output voltage limit. Simplistically stated, if theinput times the gain exceeds the voltage rail, the output waveform isgoing to be distorted.

There are two common methods of addressing the problem of the inputtimes the gain exceeding the voltage rail: clipping or compressing.Clipping refers to the output wave form being flattened anytime theinput times the gain exceeds the voltage rail. Compression refers to acontrol mechanism, within an audio amplifier or processor, that lowersthe system gain so that the gain times the input does not exceed theoutput. Compression is done in real-time in many audio amplifiers,meaning that the gain if varying in real time. As such, Compression is alinear distortion. Although Compression does not affect the TotalHarmonic Distortion of a typical audio system, it is, nonetheless,perceptible, and objectionable, to the end-user. Compression is used inalmost all audio amplifiers, with clipping being used in only the mostinexpensive and lowest performing systems.

Power Compression is an artifact created by the topology andconstruction of most audio systems. Loudspeakers (the electrical“Load”), be they floor-standing, bookshelf, desktop, or headphone,typically are current-dependent devices, meaning that they followLorentz' Force Law: F=Bxli. The force (F), driving the diaphragm, isdependent on the current (i) through a length of wire (I), that isorthogonal to the magnetic field (B).

Since most electric-to-audio transducers follow Lorentz' Law, and sincemost voltage-controlled amplifiers are limited by a voltage rail,dynamic range and compression become concerns for most audio systems. Auseful definition of Dynamic Range is the ratio of the maximumundistorted sine wave to the noise floor, which is the level at which auseful audio signal subsides into the ambient noise of the system. Audiosystem limitations with respect to Dynamic Range can be measured byCrest Factor. Crest Factor is the ratio between the r.m.s. and themaximum undistorted voltage values in an audio system. From an end-userstandpoint, a larger Dynamic Range is preferable. Simultaneously, at aparticular listening-level (voltage), a higher Crest Factor ispreferable from an end-user standpoint. Diminuendo Factor, orDecrescendo Factor, is defined in this invention as the ratio betweenthe r.m.s. level and noise floor. For a particularly idealized system,the Diminuendo Factor plus Crest Factor, as measured in dB, should equalthe Effective Dynamic Range. The Effective Dynamic Range, or theDiminuendo plus Crest Factor, are limited by the voltage rails or byproblems inherent in the dynamics processor, itself. The audio volumeinformation can only be presented as voltage variations between thenoise floor and the voltage rail, which is the Effective Dynamic Range.

There are many different algorithms for implementing a compressionstrategy. Mostly, compression uses a limiter circuit, or a DSP whichmimics a limiter circuit, and lowers the Crest Factor, so that theEffective Dynamic Range (Diminuendo Factor plus Crest Factor) is withinthe voltage rails. Since compression, as currently implemented, affectsmostly the Crest Factor, and not the Diminuendo Factor, it colors theoutput of the sound.

Current technology fails to give the listener the ability to control therelative values of compression, Dynamic Range, Diminuendo Factor, andCrest Factor. These values can be measured and quantified as ispresented in this system and method. Rather, the current solutions areall concerned with imposing built-in relationships between thecompression, Dynamic Range, Diminuendo Factor, and Crest Factor.Furthermore, current technology fails to allow a user to measure andanalyze the effects of signal dynamics modifier devices with respect toEffective Dynamic Range, Diminuendo Factor and Crest Factor .

For example, U.S. Pat. No. 8,194,869, by named inventors Ingalls, et.al. (“Ingalls '869”), entitled, “System and method for harmonizingcalibration of audio between networked conference rooms,” teaches amethod a power management system for an audio system in which aprocessor computes the real time parameters of the loudspeaker in theaudio system; a threshold comparator measures the parameters, andcompares them versus an estimated operation characteristic; and alimiter which can adjust the audio output signal to the amplifier inreal time, according to the operational characteristics. Ingalls '869teaches a method of varying compression, in real-time, to meet apre-determined performance criteria. Ingalls '869 is concerned with theload of the system, in other words, the loudspeaker. Ingalls '869 doesnot teach a method for adjusting the amplifier of an audio system basedoff of the compression of the system. Furthermore, Ingalls '869 does notteach any method to have the user adjust the parameters of the audiosystem in order to meet the end-user's preferred settings.

U.S. Pat. No 7,672,462, by named inventors Yamanashi, et. al.(“Yamanashi '462”), entitled, “Voice/musical sound encoding device andvoice/musical sound encoding method,” teaches a system and method toprotect against acoustic shock. Yamanashi '462 discloses a systemcomprising a method of receiving an input signal in the time domain andperforming pattern analysis of the signal. The pattern analysis includesproviding a filterbank with an oversampled signal representation, whicha processor transfor.m.s. into a plurality of band signals in thefrequency domain. Furthermore, the pattern analysis extractscharacteristics, using both a fast average and slow average methodology,in order to arrive at parameters on which to form a decision. Yamanashi'462 does not disclose any methodology which measures compression, CrestFactor, Diminuendo Factor, or Dynamic Range. Additionally, Yamanashi'462 does not include any method or means for an end-user to adjust theaudio output.

U.S. Pat. No. 7,583,137, by named inventors Pedersen, et. al. (“Pedersen'137”), entitled, “Power supply compensation,” teaches a method ofcompensating for power supply errors in switching amplifiers. In oneembodiment, Pedersen '137 discloses a method of measuring power supplyerror, creating a compensation signal, and mixing the compensationsignal with the input signal. Pedersen '137 does not address variablegain, compression, expansion, or dynamic range. Pedersen '137 does notmention user perception of compression, nor does it allow for useradjustment. Pedersen '137 does not disclose any methodology whichmeasures compression, Crest Factor, Diminuendo Factor, or Dynamic Range.Lastly, Pedersen '137 does not include any method or means for anend-user to adjust the audio output.

U.S. Pat. No. 6,621,338, by named inventor Van Schyndel (“Van Schyndel'338”), entitled, “Gain determination for correlation processes,”teaches a method of adjusting the maximum level of voltage given to anelectronic device (Load), by adjusting the output level, so that thesupplied output voltage exceeds the maximum level of the Load for anamount of time greater than zero. Van Schydel '338 further disclosesthat the Load can be an analog-to-digital converter. Van Schydel '338does not disclose any methodology which measures compression, CrestFactor, Diminuendo Factor, Dynamic Range, or user perception.Additionally, Van Schydel '338 does not include any method or means foran end-user to adjust the audio output.

U.S. Pat. No. 4,035,739, by named inventors Dickopp, et. al (“Dickopp'739”), entitled, “Amplifier with variable gain,” teaches a method ofvariable gain control that is frequency dependent, to overcome problemscommon with compander circuits. Dickopp '739 discloses a circuit forvariable gain control in which the lower frequency cut-off isvoltage-dependent, so that apparent noise will not beamplitude-modulated by the low-frequency signal energy density. Dickopp'739 does not disclose any methodology which measures compression, CrestFactor, Diminuendo Factor, or Dynamic Range. Additionally, Dickopp '739does not include any method or means for an end-user to adjust the audiooutput.

U.S. Pat. No. 8,229,125, by named inventor Short (“Short '125”),entitled, “Adjusting dynamic range of an audio system,” teaches anautomatic gain control method that has three inputs: output signaldynamic range, maximum output signal level, and minimum output signallevel. The user must specify one of the three quantities. The secondparameter is determined based off of the first, user-defined, parameter.The input is segmented, in the frequency domain, into frequency bands.The gain is adjusted in each band based off of the first and secondparameters. Short '125 does not disclose any method or apparatus whichaccounts for the users perceptual preference. Furthermore, Short '125does not disclose any relationship between the compression, DynamicRange, the Crest Factor, and the Diminuendo Factor, which allows the enduser to select the user's preferred point for the values. Lastly, Short'125 does not teach any user viewable measurement or analysis thatcompares compression, Dynamic Range, the Crest Factor, and theDiminuendo Factor.

U.S. Pat. No. 8,326,444, by named inventors Classen, et. al. (“Classen'444”), entitled, “Method and apparatus for performing audio ducking,”teaches a system and method for performing audio ducking between two ormore signal. The first signal is analyzed, and its characteristic data,defined as its average level, is maintained for further analysis. Asecond signal is ducked with the first signal, by adjusting the secondsignal based on the characteristic data of the first signal. Classen'444 does not disclose any methodology which measures compression, CrestFactor, Diminuendo Factor, or Dynamic Range. Additionally, Classen '444does not include any method or means for an end-user to adjust the audiooutput. Classen '444 is concerned solely with ducking two or moresignals together based on relative levels.

U.S. Pat. No. 8,300,849, by named inventors Smirnov et. al. (“Smirnov'849”), entitled, “Perceptually weighted digital audio levelcompression,” teaches a digital audio method by taking an input signal;dividing it into frequency blocks; measuring its loudness using aperceptual filter within each block; determining its gain based ontarget loudness level and measure of loudness within each block; anddetermining a frequency-dependent gain amount using piecewise linearattack/release logic. Smirnov '849 does not teach a method for useradjustment of compression levels, nor does Smirnov '849 teach a methodfor determining Crest Factor, Lower Limit Tolerance, Clip Level, andallowing the user to adjust the same.

U.S. Pat. No. 7,848,531, by named inventors Vickers et. al. (“Vickers'531”), entitled, “Method and apparatus for audio loudness and dynamicsmatching,” teaches a method of defining and adjusting apparent loudness.An audio processor, used in conjunction with a compressor, divides thesignal into time frames; determines apparent loudness within each timeframe; weights the frames to emphasize louder frames, while includingcontribution of less loud frame; and adjusts the loudness of track basedon calculation via nonlinear compression transfer function. Vickers '531does not teach a user-defined adjustment of compression parameters, suchas Crest Factor, Diminuendo Factor, and Clip Tolerance. Neither doesVickers '531 teach any method of comparing a pre- and post-processeddynamics signal.

The current prior art fails to provide a user-adjustable method ofsetting the level of audio dynamics, such as Effective Dynamic Range,Crest Factor, Diminuendo Factor, and Clip Level. Furthermore, thecurrent prior art fails to provide adequate tools to measure and comparesuch factors, including a comparison of the input and output of adynamics modifying device.

SUMMARY OF THE INVENTION

The present invention seeks to overcome the limitations in the priorart, and consists of a novel analysis and measurement technique for theevaluation of processors which modify signal dynamics. This new approachutilizes statistical analysis techniques to provide a direct comparisonand evaluation between the processed signal and the unprocessed signal'sdynamic characteristics, these characteristics being defined in thisdocument. Implementations of this method include, at a minimum, a DSP orother processor with associated software, a means of collecting data, ameans for measuring system dynamics, and a means of adjusting systemdynamics. The present invention can be part of a measurement andanalysis tool, such as an analyzer; part of an audio system, such as inan integrated audio processor and amplifier; or a stand-alone tool,which can be introduced into the signal chain.

A Probability Density Function shows a histogram of the signal peaks orroot means square (r.m.$) values versus a decibel scale, usually with 0decibels at the far right, and negative decibels being shown going leftfrom the 0 db mark. Digitized time varying signals comprise a series oftime-marked absolute values. For example, the well-known Nyquist ruledictates that the rate of sampling of continuous time variant signalsmust be twice the frequency of the highest frequency of interest. Inaudio systems this indicates a sampling frequency in excess of twice thehighest audible frequency which is considered to be 20 KHz. Hencesampling frequencies of 44.1 KHz and 48 KHz are common. At a samplingrate of 48 KHz, 48,000 data values are captured every second. This meansthat a three minute musical piece, in digitized form, is represented byapproximately 8.64 million data values. This provides a very largedatabase of numerical values ripe for new forms of statistical analysis.

This method addresses the need to measure the effect of a dynamicsmodifying device on the actual range of levels occurring in the programsignal with a recognition of the impact of time considerations such asduration of crescendi and diminuendi, the length of the passage or pieceand the duration and nature of extreme levels or peaks. The method isbased in analysis of probability densities for a number ofcharacteristics which can be derived from the signal data stream.

In order to fully illustrate the present invention, some technicaldiscussion and definitions are helpful. An inflection point occurs whenthe sequence of data values in the data stream reverses direction. Asignal peak event occurs when the inflection point changes from anincreasing sequence to a decreasing sequence. Most often such aninflection point will be a single point of data indicating a momentarypeak in the modulus of signal absolute value. Less often several datavalues in sequence might remain equal during an inflection. This wouldmost commonly occur where the signal is clipped, meaning that thetransmission medium's maximum amplitude has been reached or some signalhandling or processing algorithm has overrun its data capability. Morerarely this may be a natural aspect of the source of the signal such asa musical instrument which creates very high harmonic content. Theseinflection points represent the edges of the envelope of the signal overtime. A collection of the values of all inflection points, or peaks, ina single piece allow a probability density to be computed to representthe range in this envelope over time. See, for example, FIG. 2 whichshows an example of a distribution function for signal peaks 111. If theupper extent of the range of inflection point values has a highprobability density value 110, this indicates that dynamic range isbeing lost or constrained by the action of dynamics processing or bylimitations in the transmission medium's maximum level capability.

The highest data value in the entire distribution is the highest peak tooccur during the passage under examination. However, its probabilitydensity may be so low as to render it statistically insignificant. Formeaningful measures of the peaks in a passage it is useful to set aminimum probability density and determine the upper limit signalabsolute value for that probability density. See, for example, FIG. 9.This computed maximum (peak) value 23, based on the user's set minimumsignificant probability density 21 can then be used as a reference orbenchmark in analysis of all other computed signal data. An additionalconsideration in setting this reference upper bound is the end-user'stolerance for clipping events in the received signal. In audio listeningtests some degree of clipping is well tolerated and often preferred asproviding a greater perception of loudness or impact. Hence thisselected upper limit will be defined as “Clip Tolerance” or CT expressedas the level corresponding to a selected percentage of the totalpopulation of samples which may be clipped without a perceived reductionin information or listener satisfaction and from which a maximum peaklevel at that Clip Tolerance point is derived.

Computation of root mean square value necessarily involves multiple datapoints. In time variant signals these multiple data points comprise asequence of samples occurring in a period of time, a time window(defined here as Tr.m.s.). Short term r.m.s. computations can beaccumulated and their probability densities computed to indicate thevariation and range in volume loudness through the piece. Choice of thecomputational time can be varied according to the analytical objective.In audio signals a time constant on the order of the half-period of thelowest frequency of interest is often useful.

As with the case of signal inflection points (peaks), if the upperextent of the range of the short term r.m.s. values has a highprobability density value, observable as a truncation of thedistribution curve by steep discontinuity, this indicates that dynamicrange is being lost or constrained by the action of dynamics processingor by limitations in the transmission medium's maximum level capability.See FIG. 2, showing the upper bound curtailment 110. On the other hand,if the lower extent of the range of short term r.m.s. values has a highprobability density value, also observable as a truncation of thedistribution curve by a steep discontinuity, this indicates that dynamicrange is being lost or constrained by the action of the dynamicsprocessing or by limitations in the transmission medium noise floor. SeeFIG. 11, showing lower bound curtailment 61.

An additional limit of interest is the lowest level of perceivedinformation the listener or recipient can receive, which is defined asthe Lower Limit Tolerance (“LLT”), corresponding to the recipient'snoise floor. FIG. 10 shows an r.m.s. signal probability densitydistribution 118. A user defined minimum significant probability densityvalue 20 is shown, corresponding to the LLT. The intersection of LLT 20with the probability density distribution 118 provides an r.m.s. levelregarded as the lowest level 219 of interest in computing DiminuendoFactor. This level of zero perceived information could be due tosubjective factors such as listener preference or physiological factorssuch as hearing loss or environmental factors such as environmentalnoise as experienced in moving a car or around industrial machinery.

Where the time window is long, for example over an entire passage ormusical piece, the Long Term Root Mean Square (LT_(r.m.s.)) value willrepresent the average level throughout the segment, passage or entiremusical piece. In this method, the long term r.m.s. time represents thetotal time segment subject to analysis, whether this is the entirety ofa musical piece or a set time window (several seconds typically)allowing the analysis to be repeated continuously during the duration ofprogram material (test signal or music). The upper limit of theLT_(r.m.s.) time window is the total time for the complete passage ormusical piece's r.m.s. computation denoted T_(pop).

Each of the peak and r.m.s. probability density measurements, computedusing the user's set values for CT, LLT, and LT_(r.m.s.), can bedivided, or filtered, in the frequency domain to provide multiple,frequency-dependent, Probability Density Functions across, the frequencyspectrum of the signal analyzed in frequency bands. A passage of music,for example, may have quite different Probability Density Functions inthe low-frequency range versus the mid- and the high-frequencies.Analysis and adjustment, using the presently described technique, wouldthen be used to examine and optimize the differences in the effect ofthe dynamics modifier device across the frequency spectrum. Differencesin the degree of dynamics modification across the frequency spectrum aregenerally perceived as unsatisfactory in listening tests, but are,often, deliberately applied for artificial effects or for systemoverload protection. As an example low frequencies might be compressedmore aggressively than high frequencies as low frequencies tend to havemore energy than higher frequencies in natural sounds indicating agreater danger of system overload from lower frequencies. In speechreproduction systems it can be desirable to preserve the inherentdynamics of the middle and higher frequencies to preserveintelligibility while compressing and limiting lower frequencies toavoid overload.

The Probability Density Functions of these parameters, as describedabove, will typically have the characteristics of a skewed bell curvewhere the peak of the curve represents the median value. As noted adiscontinuity in the form of a curtailment of the skirt of thedistribution curve indicates a limitation of the transmission medium: inthe case of an upper limitation, a maximum capability of the medium isreached called clipping or overload; and, in the case of a lowerlimitation the noise floor of the medium is the restriction on thelowest level of usable signal.

The actual Dynamic Range equals the difference between the maximum leveland the noise floor. However, the Effective Dynamic Range is thedifference between CT and LLT. The present invention includes means forthe end-user to set both CT and LLT, allowing, for the first time, theuser to define the Effective Dynamic Range. As will be discussed, thishas an impact on perceived loudness, also.

Crest factor is commonly defined as the ratio of the maximum peak signallevel to the r.m.s. level during a specified time window. Unfortunatelythe maximum peak signal is poorly defined. Where the maximum attainedvalue occurs only once through the section being analyzed this peakevent is not statistically significant and results in a crest factorvalue of little utility. However, in a time window in which clipping isprevalent, the upper bound of the crest factor will be the voltage rail.The present invention defines a variant of Crest Factor as being theratio of the CT level (or the system maximum level where upper boundcurtailment due to clipping is evident at a level below the CT level) tothe median r.m.s. value . We can then further define Diminuendo Factoras the ratio of the median r.m.s. value to the level corresponding tothe LLT or the Noise Floor (whichever is higher).

In addition the ratio of the Diminuendo Factor relative to the CrestFactor, which will be referred to as the Relative Loudness, provides anindication of the perceived loudness of the signal program content.Where a dynamics modifier device alters the relationship betweenDiminuendo Factor and Crest Factor the perceived loudness is affectedand alters the listeners experience more than a situation where theRelative Loudness relationship is preserved.

Each probability density function, be that from a population of peakdata (inflection points) or short term r.m.s. value computations, can bedescribed in terms of standard statistical terms including, but notlimited to, median, average, standard deviation, skew and kurtosis.

This invention is a method to evaluate, and, potentially, adjust, thedynamics modifier devices in terms of their effect on the definedprobability distributions, before and after the dynamics modifyingdevice. In essence the method provides ratios of the post-processed andoriginal (input) signal's characteristics such as Effective DynamicRange, Crest Factor, Diminuendo Factor, Relative Loudness, Median r.m.s.level, Skew and Kurtosis. The operator sets values for Clip Tolerance(CT), Low Level Tolerance (LLT), Short term r.m.s. time window(T_(r.m.s.)), and Long term r.m.s. time window (LT_(r.m.s.)) to enablemeaningful ratios to be computed from the post processed and originalsignal's Probability Density Functions. Additionally, this inventiongives the end-user, typically the listener or system-design engineer,the ability to adjust relative levels of Crest Factor and DiminuendoFactor for a given Effective Dynamic Range. This allows the end-user tochoose, in more or less a continuous fashion, the perceived RelativeLoudness of the system.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 shows a system architecture diagram of the present invention,using an embodiment that allows user measurement and analysis of adynamics-modifying device, such as a processor, and the interaction tovarious pre- and post-processed parameter ratios.

FIG. 1B shows a system architecture diagram of the present invention,using an embodiment that allows the user the ability to measure,analyze, and adjust a dynamics-modifying device, such as a processor,and the parameters and interactions that can be examined and adjusted.

FIG. 2 shows a representative chart or graph of a Probability DensityFunction of signal peaks with upper bound curtailment.

FIG. 3 shows a representative chart or graph of the Probability DensityFunction of pre- and post-processed median r.m.s. values, taken overLT_(r.ms.) (or T_(pop), the upper limit of LT_(r.m s.)), and theirrelationship to pre-processed diminuendo and crest factors.

FIG. 4 shows a representative chart or graph of the Probability DensityFunctions for the original r.m.s. signal voltage and compressed voltageoutput values, taken over LT_(r.m.s.), in which lower bound curtailmentis present.

FIG. 5 shows a chart of Probability Density Function for r.m.s. voltageoutput values, taken over LT_(r.m.s.), and its relationship to LLT,Crest Factor, and Clip Tolerance Level.

FIG. 6 shows a chart of Probability Density Function for r.m.s. voltageoutput values, taken over LT_(r.m.s.), and its relationship toDiminuendo Factor and Crest Factor.

FIG. 7 shows a chart of Probability Density Function for r.m.s. voltageoutput values, taken over LT_(r.m.s.), and its relationship to EffectiveDynamic Range.

FIG. 8 shows a chart of Probability Density Function for the originalr.m.s. and compressed voltage output values, taken over LT_(r.m.s.),with the resultant curves normalized to the mean level to expose thechanged nature of the distribution curve tails.

FIG. 9 shows an example of a Probability Density Function chart for peakvoltages, highlighting the upper bound of the minimum significantprobability and its relationship to Clip Tolerance.

FIG. 10 shows an example of a Probability Density Function, highlightingthe relationship between the lower level signal tolerance (LLT), and thelower bound for the minimum significant probability density.

FIG. 11 shows a Probability Density Function chart for r.m.s. value orvoltage, measured over LT_(r.m.s.), showing lower bound curtailmentcaused by the noise floor.

FIG. 12 shows a chart relating Probability Density Function of pre- andpost-processed median r.m.s. values and their relationship topost-processed Diminuendo and Crest factors.

Note that for FIGS. 2 through 12 the horizontal axis is signal level indB and the vertical axis is unscaled probability.

DETAILED DESCRIPTION AND SUMMARY OF THE INVENTION

This description does not limit the invention, but rather illustratesits general principles of operation. Examples are illustrated with theaccompanying drawings. A variety of drawings are offered, showing thepresent invention and the probability density functions that are used inits algorithm.

A Probability Density Function for audio input or output voltage,whether peak or r.m.s., shows a sequence of data values that, ideally,monotonically increase, until a maximum, and then monotonicallydecrease, until a minimum.

When the sequence of signal absolute value data values in the datastream reverses direction, it indicates an inflection point. When theinflection point in the absolute value changes from an increasingsequence to a decreasing sequence this indicates a signal peak event.Most often, such an inflection point will be a single point of dataindicating a momentary peak in the modulus of signal absolute value, butit can also be a sequence of values, when the signal is clipped.

FIG. 1 shows the present invention, along with a probability densityanalysis chart. The invention relies on an input signal 1. The signal ispassed through a dynamics processor 2, which is the device under test(“DUT”) 2. The DUT 2 is any device that has a processor that can affectthe signal dynamics: compression, expansion, dynamic range, or clippingof a signal. The DUT 2 can include, but is not limited to, a DSP, asignal processor, an integrated amplifier with a DSP that has acompressor, a computer with sound card and software, or an audio system.The DUT 2 outputs an output signal 3. The invention has a processor thatcan compare the pre-processed probability density 4 of the signal andthe post-processed probability density 5 of the signal. The processorcan received value settings for root mean square time constant, T_(rms),6; long-term root mean square time window, LT_(rms), 7; totalperformance time, T_(pop), 8; Clip Tolerance, CT, 9; and Low LevelTolerance, LLT, 10.

The Probability Density Function pair 11 shows the relationship betweena typical pre-processed 13 and post-processed 14 Probability DensityFunctions. The processor can easily compare the pre-processed 4 andpost-processed 5 Probability Density Functions. The processor can assessa number of parameters and ratios 16, between pre- and post-processedvalues, including, but not limited to, Dynamic Range, Crest Factor,Diminuendo Factor, Relative Loudness, Median r.m.s., Kurtosis, and Skew.

The median value to the noise floor is the Diminuendo Factor 15. In thisdrawing, the Diminuendo Factor 15 is for the post-processed medianr.m.s. value. The post-processed Crest Factor 12 is the CT Level to themedian r.m.s. value (corresponding to the peak of the probabilitydensity distribution) of the Probability Density Function.

FIG. 1B shows an alternative embodiment for this analysis apparatus usedto evaluate the effect of power compression in a loudspeaker system. Inthis case the output signal 103 is derived from an acoustic signal 106,which is the ouput of the DUT 109. In this embodiment, the DUT 109 is anelectro-acoustic system 109, composed of, at a minimum, an amplifier 108and a loudspeaker 107. The DUT receives an input 1. A measurementmicrophone 105 is connected to a measurement preamplifier 104 to providea signal to the post compression probability density analysiscomputation 5. The measurement output signal 103 shows the result ofsuch system. Several measurements are consistent between FIG. 1 and FIG.1B: the pre-processed Probability Density Function 4, T_(rms) 6, LTrms7, Tpop 8, Clip Tolerance 9 or CT 9, and Low Level Tolerance 10 or LLT10. 0045 FIG. 2 shows an example of a Probability Density Function 112chart for signal peak levels 111 in which the upper level curtailmenttypical of overload or clipping 110 is evident in a sharp discontinuityin the upper tail of the curve 110. The curve shows a sharp upwarddiscontinuity 110 caused by repeated level events at the system maximumwhich, absent this level constraint, would occur at higher levels thanthe level at 110.

FIG. 3 shows a graph 11 between pre-13 and post-processed 14 ProbabilityDensity Functions 13, 14. The relationship between the pre-processedDiminuendo Factor 18 and pre-processed Crest Factor 19 is illustrated.

FIG. 4 shows a Probability Density Function graph 11, comparing theoriginal signal r.m.s. 118 and compressed 17 Probability DensityFunction. The tail 119 of the original r.m.s. and the compressed 17curves is shown for an example of lower bound curtailment due systemconstraints caused by the system noise floor.

FIG. 5 shows an r.m.s. Probability Density Function graph 118,specifically calling out its median level as the peak of the curve. Theminimum significant Probability Density Function (upper) 21 defined asClip Tolerance probability value 23 and minimum significant ProbabilityDensity Function (lower) 20 defined as Lower Limit Tolerance probabilityvalue 219 are shown. The level at Clip Tolerance 23 is obtained from theintersection of peak level Probability Density Function curve 51 of FIG.9 and the minimum significant probability density (upper) 21. The CrestFactor 22 is the difference between the Clip Tolerance Level (CT) 23 andthe median r.m.s. level (peak of 118). The Lower Level Signal Tolerance(LLT) 219 is the intersection of the r.m.s. Probability Density Functioncurve 118 and the minimum significant probability density (lower) 20,which is usually considered the noise floor.

FIG. 6 shows an r.m.s. Probability Density Function 118, specificallycalling out its median level as the peak of the curve. The minimumsignificant Probability Density Function (upper) 21 defined as ClipTolerance probability value 23 and minimum significant probabilitydensity (lower) 20 defined as Lower Limit Tolerance (LLT) value 219 areshown. The level at Clip Tolerance 23 obtained from the peak levelProbability Density Function 51 of FIG. 9 is shown, and, as the peaklevel corresponding to the minimum significant probability density(upper) 21. The Crest Factor 22 is the difference between the ClipTolerance Level (CT) 23 and the median r.m.s. level (peak of 118). TheLower Level Signal Tolerance (LLT) 219 is the intersection of the r.m.s.probability density 118 and the minimum significant probability density(lower) 20. The Diminuendo Factor 24 is the difference between theMedian r.m.s. level peak (callout of 118) and the Lower Level SignalTolerance (LLT) 219.

Comparing FIG. 6 to FIG. 7 shows that the Effective Dynamic Range 25(FIG. 7) is equal to the Diminuendo Factor 24 plus Crest Factor 22(FIG6).

FIG. 8 shows the same curves as FIG. 12, being original signal r.m.s.Probability Density Function 26 compared to a compressed probabilitydensity function 27. In this chart, the curves have been normalized tothe median level to visually highlight the characteristic of the tails28, 29 of each curve in which the compressed curve tails show morekurtosis, i.e. steeper tails, than the original signal curve.

FIG. 9 shows an example Probability Density Function chart for signalpeak levels in which the minimum significant probability density (upper)for highest level signal peaks considered statistically significant hasbeen set 21. This Clip Tolerance 23 probability corresponds to theintersection of the peaks curve 51, and the minimum significantprobability density 21.

FIG. 10 shows an example of a Probability Density Function for r.m.s.

voltage value 118, using a T_(rms) time window, highlighting the LLT 219and the minimum significant probability density (lower) 20. The LLT 219is the intersection of the lower portion fo the r.m.s. ProbabilityDensity Function 118 and the minimum significant probability density(lower) 20. The minimum significant probability density (upper) 21 isalso shown for reference.

FIG. 11 shows a curve 63 of a Probability Density Function 62 for thesignal short term r.m.s, taken over time value LT_(rms). The lowerportion of the curve shows the curtailment of the lower bound at thenoise floor 61.

FIG. 12 shows a comparison between pre- 13 and post-processed 14Probability Density Functions 11. The relationship between thePost-processed Diminuendo Factor 15 and Post-processed Crest Factor 12is apparent. Similarly the same quantities could be calculated for thePre-processed values, as shown in FIG. 3.

I claim:
 1. A method for allowing a user to adjust the compressionlevels of an audio signal, the method comprising: an audio signal input;an audio output signal; a means of measuring the probability densityfunction of a signal for a given time window; a processor that cancalculate the Effective Dynamic Range (difference between Clip Toleranceand Lower Limit Tolerance) and the Crest Factor (ratio of the mediansignal value to the Clip Tolerance), and the Diminuendo Factor (ratio ofthe median signal value to the LLT); and a means for control that allowsfor an adjustment of the sequence of voltage values to alter therelative relationship of Crest Factor, Effective Dynamic Range,Diminuendo Factor, and Lower Limit Tolerance within total frequencybands of interest (e.g., the audio band from 20-20 kHz) or a sub-band orsub-bands of the frequency band of interest, such as the bass range orthe middle frequency range; or multiple sub-bands analyzed and processedseparately and simultaneously.
 2. The invention described in 1, in whichthe means for control further comprises a method that can be used as anintegral element within the control loop of a dynamics processor toallow the user to set target values for the outcomes for both absolutecharacteristics, in particular, two out of: Effective Dynamic Range,Effective Crest Factor and Diminuendo Factor, and relativecharacteristics in particular the ratios of Relative Loudness, Skew,Kurtosis, and Median r.m.s.
 3. The invention described in 1, in whichthe means of setting user-defined levels for Clip Tolerance and LowerLimit Tolerance is further comprised as a control which allows the userto adjust the relationship between CT, LLT, Diminuendo Factor, andEffective Dynamic Range, such as a voltage-controlled potentiometer, ora digital means of mimicking such a potentiometer, including the digitaloutput from a rotary encoder, the output of a digital counterincremented using up and down buttons or from graphical representationsof such controls within a graphical user interface in a computer device(desktop, laptop, tablet, smartphone etc.).
 4. The invention describedin 1, further comprising a means for setting user-defined levels forShort Term Time Window (T_(rms)), and Long Term Time Window (LT_(rms)){or its upper limit value T_(pop)}, in which the user uses apotentiometer that linearly scales the time window, or a digital meanswhich mimics such a potentiometer, in order to adjust the T_(rms),LT_(r.m.s.) and T_(pop).
 5. The invention described in 1, furthercomprising a means for displaying the Effective Dynamic Range, ClipTolerance, Lower Limit Tolerance, Crest Factor, and Diminuendo Factor.6. A apparatus for allowing a user to adjust the compression levels ofan audio signal, the apparatus comprising: an audio input; an audiooutput; a processor that can calculate the Effective Dynamic Range(difference between Clip Tolerance and Lower Limit Tolerance), the CrestFactor (ratio of the median signal value to the Clip Tolerance), and theDiminuendo Factor (ratio of the median signal value to the LLT); and ameans for setting user defined levels for Clip Tolerance, and LowerLimit Tolerance.
 7. The invention described in 6, further comprising ameans for setting user-defined levels for Short Term Time Window(Tr.m.s.), and Long Term Time Window (LT_(r.m.s)) {or its upper limitvalue T_(pop)}.
 8. The invention described in 6, further comprising ameans for displaying the Effective Dynamic Range, Clip Tolerance, LowerLimit Tolerance, Crest Factor, and Diminuendo Factor.
 9. A system forallowing a user to adjust audio compression levels, the systemcomprising: an electrical-to-audio transducer; a device for processingthe Effective Dynamic Range (difference between Clip Tolerance and LowerLimit Tolerance), the Crest Factor (ratio of the median signal value tothe Clip Tolerance) and the Diminuendo Factor (ratio of the mediansignal value to the LLT), said device having an audio input and an audiooutput; an amplifier for raising the voltage of the output signal intothe transducer to a user-definable level; and a means for settinguser-defined levels for Clip Tolerance, and Lower Limit Tolerance. 10.The invention described in 9, further comprising a means for settinguser-defined levels for Short Term Time Window (Tr.m.s.), and Long TermTime Window (LT_(r.m.s)) {or its upper limit value T_(pop)}.
 11. Theinvention described in 9, further comprising a means for displaying theEffective Dynamic Range, Clip Tolerance, Lower Limit Tolerance, CrestFactor, and Diminuendo Factor.